Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 364 matching lines @@
         webrtc::Trace::ReturnTrace();
         exit(1);
     }
   }
   return payload_len;
 }
 
 }  // namespace
 
 int main(int argc, char* argv[]) {
-  static const int kMaxChannels = 5;
-  static const size_t kMaxSamplesPerMs = 48000 / 1000;
   static const int kOutputBlockSizeMs = 10;
 
   std::string program_name = argv[0];
   std::string usage = "Tool for decoding an RTP dump file using NetEq.\n"
       "Run " + program_name + " --helpshort for usage.\n"
       "Example usage:\n" + program_name +
       " input.rtp output.{pcm, wav}\n";
   google::SetUsageMessage(usage);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
@@ skipping 202 matching lines @@
       } else {
         // Set next input time to the maximum value of int64_t to prevent the
         // time_now_ms from becoming stuck at the final value.
         next_input_time_ms = std::numeric_limits<int64_t>::max();
         packet_available = false;
       }
     }
 
     // Check if it is time to get output audio.
     while (time_now_ms >= next_output_time_ms && output_event_available) {
-      static const size_t kOutDataLen =
-          kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
-      int16_t out_data[kOutDataLen];
-      size_t num_channels;
-      size_t samples_per_channel;
-      int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
-                                   &num_channels, NULL);
+      webrtc::AudioFrame out_frame;
+      int error = neteq->GetAudio(&out_frame, NULL);
       if (error != NetEq::kOK) {
         std::cerr << "GetAudio returned error code " <<
             neteq->LastError() << std::endl;
       } else {
-        // Calculate sample rate from output size.
-        sample_rate_hz = rtc::checked_cast<int>(
-            1000 * samples_per_channel / kOutputBlockSizeMs);
+        sample_rate_hz = out_frame.sample_rate_hz_;
       }
 
       // Write to file.
       // TODO(hlundin): Make writing to file optional.
-      size_t write_len = samples_per_channel * num_channels;
-      if (!output->WriteArray(out_data, write_len)) {
+      if (!output->WriteArray(out_frame.data_, out_frame.samples_per_channel_ *
+                                                   out_frame.num_channels_)) {
         std::cerr << "Error while writing to file" << std::endl;
         webrtc::Trace::ReturnTrace();
         exit(1);
       }
       if (is_rtp_dump) {
         next_output_time_ms += kOutputBlockSizeMs;
         if (!packet_available)
           output_event_available = false;
       } else {
         next_output_time_ms = event_log_source->NextAudioOutputEventMs();
         if (next_output_time_ms == std::numeric_limits<int64_t>::max())
           output_event_available = false;
       }
     }
   }
   printf("Simulation done\n");
   printf("Produced %i ms of audio\n",
          static_cast<int>(time_now_ms - start_time_ms));
 
   delete neteq;
   webrtc::Trace::ReturnTrace();
   return 0;
 }